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authorDan Halbert <halbert@halwitz.org>2018-06-14 18:47:40 -0400
committerDan Halbert <halbert@halwitz.org>2018-06-14 18:47:40 -0400
commite724bc1c4eb1224aa043a1489b29ec3daaa832aa (patch)
treeaa72de23055ef2fb3eae33d11e1fc5536a5eb05c /ports
parent618943d90a958d36d9b0e2206552100069b10ddb (diff)
Fix playing audio from SD card
Diffstat (limited to 'ports')
-rw-r--r--ports/atmel-samd/Makefile6
-rw-r--r--ports/atmel-samd/audio_dma.c36
2 files changed, 33 insertions, 9 deletions
diff --git a/ports/atmel-samd/Makefile b/ports/atmel-samd/Makefile
index 1a406c1d0..8d454a4f6 100644
--- a/ports/atmel-samd/Makefile
+++ b/ports/atmel-samd/Makefile
@@ -102,8 +102,10 @@ endif
ifeq ($(DEBUG), 1)
# Turn on Python modules useful for debugging (e.g. uheap, ustack).
CFLAGS += -ggdb
-## CFLAGS += -flto
- CFLAGS += -fno-inline -fno-ipa-sra
+ # You may want to disable -flto if it interferes with debugging.
+ CFLAGS += -flto
+ # You may want to enable these flags to make setting breakpoints easier.
+## CFLAGS += -fno-inline -fno-ipa-sra
ifeq ($(CHIP_FAMILY), samd21)
CFLAGS += -DENABLE_MICRO_TRACE_BUFFER
endif
diff --git a/ports/atmel-samd/audio_dma.c b/ports/atmel-samd/audio_dma.c
index 5a572e9d1..4ac6079e9 100644
--- a/ports/atmel-samd/audio_dma.c
+++ b/ports/atmel-samd/audio_dma.c
@@ -34,6 +34,11 @@
#include "py/mpstate.h"
+static audio_dma_t* audio_dma_state[AUDIO_DMA_CHANNEL_COUNT];
+
+// This cannot be in audio_dma_state because it's volatile.
+static volatile bool audio_dma_pending[AUDIO_DMA_CHANNEL_COUNT];
+
uint32_t audiosample_sample_rate(mp_obj_t sample_obj) {
if (MP_OBJ_IS_TYPE(sample_obj, &audioio_rawsample_type)) {
audioio_rawsample_obj_t* sample = MP_OBJ_TO_PTR(sample_obj);
@@ -70,7 +75,7 @@ uint8_t audiosample_channel_count(mp_obj_t sample_obj) {
return 1;
}
-void audiosample_reset_buffer(mp_obj_t sample_obj, bool single_channel, uint8_t audio_channel) {
+static void audiosample_reset_buffer(mp_obj_t sample_obj, bool single_channel, uint8_t audio_channel) {
if (MP_OBJ_IS_TYPE(sample_obj, &audioio_rawsample_type)) {
audioio_rawsample_obj_t* sample = MP_OBJ_TO_PTR(sample_obj);
audioio_rawsample_reset_buffer(sample, single_channel, audio_channel);
@@ -81,7 +86,10 @@ void audiosample_reset_buffer(mp_obj_t sample_obj, bool single_channel, uint8_t
}
}
-bool audiosample_get_buffer(mp_obj_t sample_obj, bool single_channel, uint8_t channel, uint8_t** buffer, uint32_t* buffer_length) {
+static audioio_get_buffer_result_t audiosample_get_buffer(mp_obj_t sample_obj,
+ bool single_channel,
+ uint8_t channel,
+ uint8_t** buffer, uint32_t* buffer_length) {
if (MP_OBJ_IS_TYPE(sample_obj, &audioio_rawsample_type)) {
audioio_rawsample_obj_t* sample = MP_OBJ_TO_PTR(sample_obj);
return audioio_rawsample_get_buffer(sample, single_channel, channel, buffer, buffer_length);
@@ -90,7 +98,7 @@ bool audiosample_get_buffer(mp_obj_t sample_obj, bool single_channel, uint8_t ch
audioio_wavefile_obj_t* file = MP_OBJ_TO_PTR(sample_obj);
return audioio_wavefile_get_buffer(file, single_channel, channel, buffer, buffer_length);
}
- return true;
+ return GET_BUFFER_DONE;
}
static void audiosample_get_buffer_structure(mp_obj_t sample_obj, bool single_channel,
@@ -117,7 +125,6 @@ uint8_t find_free_audio_dma_channel(void) {
return channel;
}
-audio_dma_t* audio_dma_state[AUDIO_DMA_CHANNEL_COUNT];
void audio_dma_convert_signed(audio_dma_t* dma, uint8_t* buffer, uint32_t buffer_length,
uint8_t** output_buffer, uint32_t* output_buffer_length,
uint8_t* output_spacing) {
@@ -163,8 +170,9 @@ void audio_dma_convert_signed(audio_dma_t* dma, uint8_t* buffer, uint32_t buffer
void audio_dma_load_next_block(audio_dma_t* dma) {
uint8_t* buffer;
uint32_t buffer_length;
- bool last_buffer = audiosample_get_buffer(dma->sample, dma->single_channel, dma->audio_channel,
- &buffer, &buffer_length);
+ audioio_get_buffer_result_t get_buffer_result =
+ audiosample_get_buffer(dma->sample, dma->single_channel, dma->audio_channel,
+ &buffer, &buffer_length);
DmacDescriptor* descriptor = dma->second_descriptor;
if (dma->first_descriptor_free) {
@@ -172,6 +180,11 @@ void audio_dma_load_next_block(audio_dma_t* dma) {
}
dma->first_descriptor_free = !dma->first_descriptor_free;
+ if (get_buffer_result == GET_BUFFER_ERROR) {
+ audio_dma_stop(dma);
+ return;
+ }
+
uint8_t* output_buffer;
uint32_t output_buffer_length;
uint8_t output_spacing;
@@ -180,7 +193,7 @@ void audio_dma_load_next_block(audio_dma_t* dma) {
descriptor->BTCNT.reg = output_buffer_length / dma->beat_size / output_spacing;
descriptor->SRCADDR.reg = ((uint32_t) output_buffer) + output_buffer_length;
- if (last_buffer) {
+ if (get_buffer_result == GET_BUFFER_DONE) {
if (dma->loop) {
audiosample_reset_buffer(dma->sample, dma->single_channel, dma->audio_channel);
} else {
@@ -347,6 +360,7 @@ void audio_dma_init(audio_dma_t* dma) {
void audio_dma_reset(void) {
for (uint8_t i = 0; i < AUDIO_DMA_CHANNEL_COUNT; i++) {
audio_dma_state[i] = NULL;
+ audio_dma_pending[i] = false;
dma_disable_channel(i);
dma_descriptor(i)->BTCTRL.bit.VALID = false;
MP_STATE_PORT(playing_audio)[i] = NULL;
@@ -367,8 +381,12 @@ bool audio_dma_get_playing(audio_dma_t* dma) {
// WARN(tannewt): DO NOT print from here. Printing calls background tasks such as this and causes a
// stack overflow.
+
void audio_dma_background(void) {
for (uint8_t i = 0; i < AUDIO_DMA_CHANNEL_COUNT; i++) {
+ if (audio_dma_pending[i]) {
+ continue;
+ }
audio_dma_t* dma = audio_dma_state[i];
if (dma == NULL) {
continue;
@@ -379,6 +397,10 @@ void audio_dma_background(void) {
continue;
}
+ // audio_dma_load_next_block() can call Python code, which can call audio_dma_background()
+ // recursively at the next background processing time. So disallow recursive calls to here.
+ audio_dma_pending[i] = true;
audio_dma_load_next_block(dma);
+ audio_dma_pending[i] = false;
}
}